All posts in SIP Standards

Skype’s email to me re: Skype for Asterisk

By Erik Lagerway | May 25, 2011 | asterisk, ietf, microsoft, P2P SIP, sip, SIP Standards, Skype, VoIP Standards | 1 Comment

I just received this email from Skype’s PR firm…

Hi Erik,

Here is Skype’s official comment regarding Skype for Asterisk.  You can attribute this to Jennifer Caukin, spokeswoman for Skype.

“Skype made the decision to retire Skype for Asterisk several months ago, as we have prioritized our focus around implementing the IETF SIP standard in our Skype Connect solution. SIP enjoys the broadest support of any of the available signaling alternatives by business communications equipment vendors, including Digium.  By supporting SIP in favor of alternatives, we maximize our resources and continue to reinforce our commitment to delivering Skype on key platforms where we can meet the broadest customer demand.”

Thank you,
Cassie

Call me crazy but if I have to pay to integrate Skype into my phone system, where I already have a phone service that I am happy with, why would I do that? Maybe I just want to be able to make/receive Skype calls on my SIP-enabled desk phone? If it doesn’t hit the PSTN why do I have to pay? Seems like an odd approach for a company that has a long history of working around POTS, much to the delight of their users.

Integration with SIP is great, don’t get me wrong, but it would be nice if Skype talked SIP and was ‘still’ free. Seems like a massive oversight on behalf of Skype or am I missing something?

Written by Erik Lagerway - Visit Website

Open and secure alternative to Skype

By Erik Lagerway | May 25, 2011 | asterisk, communications, ietf, IP Video, itsp, microsoft, mobile sip, mobile voip, Open Source, pbx, phone services, phone system, Secure VoIP, sip, SIP Standards, Skype, Small Medium Business in Canada, softphone, voip, VoIP for the Masses, Vonage | Comments Off

Imagine a new secure P2P (Skype like) offer that also supported SIP in the client. You could use the client software on it’s own (just like Skype) or attach it to just about any VoIP service or phone system for free.

Does it make sense for consumers?
Does it make sense for business users?
Is there room in the market?
Would you use it?

Martyn Davies chimes in…

I would use it, but as a telecom industry insider, I know that I’m not the average business user or consumer. As to whether there is room in the market, I think that depends a lot on what Microsoft do with Skype now that they own it. From a business point-of-view, their efforts are focused around OCS/Lync (and software licenses), so Skype there is not adding to their central proposition. Skype has a lot of users, but produces very little revenue, since the majority just use the free services. As a Skype competitor you would have the same problems getting to the cash.

Skype was really the first company to take VoIP and make it completely trivial to install and use. To do that, they had to take some liberties and deviate from standards (like SIP), so that they could add the magic that made it work from behind firewalls, add security and self-configuration, and integrate video so seamlessly. Like Facebook, once it is clearly the biggest of its kind of services, it becomes the community that everyone must join. I can’t see that another Skype-alike has a way in, unless Microsoft significantly change the rules now.

What do you think?

Written by Erik Lagerway - Visit Website

IETF 80 – RTCWEB BOF

By Erik Lagerway | March 28, 2011 | ietf, P2P VoIP, RTCWEB, sip, SIP Standards | Comments Off

The RTCWEB Bof meeting starts in less than 5 minutes. To some, the most anticipated meeting of IETF 80.

Proposed charter (truncated)…

The WG will perform the following work:
1.     Define the communication model in detail, including how session management is to occur within the model.
2.     Define a security model that describes the security goals and how the communication model can achieve these goals.
3.     Define how NAT and Firewall traversal is to occur.
4.     Define which RTP functions and extensions that shall be supported in the client and their usage for real-time media, including media adaptation to ensure congestion safe usage.
5.     Define what functionalities in the solution, such as media codecs, security algorithms, etc., that can be extended and how the extensibility mechanisms works.
6.     Define a set of media formats that must or should be supported by a client to improve interoperability.
7.     Define how non RTP datagram and byte stream data communication between the clients can be done securely and in a congestion safe way.
8.     Provide W3C input for the APIs that comes from the communication model and the selected components and protocols that are part of the solution.

Milestones:
Aug 2011 Architecture and Security and Threat Model sent to W3C
Aug 2011 Use cases and Scenarios document sent to W3C
Sept 2011 Architecture and Security and Threat Model to IESG as Informational
Sept 2011 Use cases and Scenarios for RTCWeb document sent to IESG as Informational
Dec 2011 RTCWeb and Media format specification(s) to IESG as PS
Dec 2011 Information elements and events APIs Input to W3C
Apr 2012 API to Protocol mapping document submitted to the IESG as Informational (if needed)

Written by Erik Lagerway - Visit Website

IETF 80 – Prague mobile roaming no workie, SIP to the rescue

By Erik Lagerway | March 27, 2011 | apple, ietf, IP Videoconferencing, iphone, mobile voip, sip, SIP Standards, softphone | Comments Off

Well, I am a little sad that I have to turn ON international mobile roaming with Bell in order to get my mobile phone working here, which it is still not, but all is not lost. I have been using FaceTime over the WifI on my MacBook Air and iPhone 4 to call my business partner and my wife back home. Kinda fitting actually, FaceTime is standards based and is all about SIP and RTP etc. Now if we can just get them to open up that API…

Keep on smiling!

Written by Erik Lagerway - Visit Website

IETF 80 Prague

By Erik Lagerway | March 27, 2011 | ietf, mobile sip, mobile voip, RTCWEB, sip, SIP Standards, voip | 2 Comments

My Chief Architect and I arrived in Prague today to attend the 80th IETF meetings. 12 hours in planes and airports makes Erik a very sad man :(

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VoIP Network Monitor ‘SIPQOS’ Launches Beta

By Erik Lagerway | February 15, 2010 | IP Communications News, itsp, pbx, phone services, phone system, SIP phone software, SIP Standards, sip trunk, small office phone systems, Startups, voip, voip nework monitor, VoIP Standards | Comments Off
Screen shot 2010-12-16 at 3.43.49 PM


Many of us have struggled with VoIP Network Monitoring, keeping tabs on our network without having to manually review the health is always a hassle and concern. For every network my team erected we needed to erect a proper monitor. For smaller networks and even VoIP phone systems the traditional Network Monitors were far to expensive to implement and required port mirroring which meant servers had to be deployed in the VoIP network that required monitoring.

So, we created SIPQOS… SIPQOS is a service that allows VoIP network administrators to attach virtual SIP endpoints to their network which send calls to-and-fro and monitors those calls for interruption. It’s a simplistic approach to a complex problem, if the network drops a registration or if a call fails it’s likely (from personal experience at least) that the issue applies to the entire network and other endpoints are experiencing the same problem. SIPQOS won’t take the place of more expensive in-network solutions but it does a great job of providing redundant VoIP network monitoring and SIP-based VoIP phone system monitoring as well.

An excerpt from the announcement we made on the 10th…

VANCOUVER, February. 10 – SIPQOS (pronounced SIP-KWOSS), a new entrant in the VoIP network monitoring market has launched a beta of its remote VoIP network monitoring service today. SIPQOS released the first product to bring the power of remote VoIP network monitoring by combining embedded SIP (Session Initiation Protocol) User Agents, web services and some secret sauce. SIPQOS monitors VoIP networks remotely and alerts network administrators when a problem has been detected.

SIPQOS is doing a great job for us and provides redundant VoIP network monitoring on a production network we run today. It also fills a void where others solutions fell flat, SMS alerts are critical and SIPQOS delivers in spades on that front. Those interested should give it a whirl, it’s free to sign up and the plans after the 30 day trial are cheap by anyone’s standards.

Written by Erik Lagerway - Visit Website
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